replace asf... by AUDIO_SAMPLE_FORMAT_... constants

git-svn-id: svn://svn.code.sf.net/p/ultrastardx/svn/branches/experimental@2769 b956fd51-792f-4845-bead-9b4dfca2ff2c

5 files changed, 28 insertions, 24 deletions

diff --git a/mediaplugin/src/mediaplugins/ffmpeg/ffmpeg_audio_convert.h b/mediaplugin/src/mediaplugins/ffmpeg/ffmpeg_audio_convert.h
index ee4c5a81..fbb5bedc 100644
--- a/mediaplugin/src/mediaplugins/ffmpeg/ffmpeg_audio_convert.h
+++ b/mediaplugin/src/mediaplugins/ffmpeg/ffmpeg_audio_convert.h
@@ -47,7 +47,7 @@ protected:
 	bool init() {
 		// Note: FFmpeg does not support resampling for more than 2 input channels
 
-		if (_srcFormatInfo.getFormat() != asfS16) {
+		if (_srcFormatInfo.getFormat() != AUDIO_SAMPLE_FORMAT_S16) {
 			logger.error("Unsupported format", "TAudioConverter_FFmpeg.Init");
 			return false;
 		}
diff --git a/mediaplugin/src/mediaplugins/ffmpeg/ffmpeg_audio_decode.cpp b/mediaplugin/src/mediaplugins/ffmpeg/ffmpeg_audio_decode.cpp
index da49891c..ce15f7a7 100644
--- a/mediaplugin/src/mediaplugins/ffmpeg/ffmpeg_audio_decode.cpp
+++ b/mediaplugin/src/mediaplugins/ffmpeg/ffmpeg_audio_decode.cpp
@@ -179,7 +179,7 @@ bool FFmpegAudioDecodeStream::_open(const IPath &filename) {
 	audioSampleFormat_t sampleFormat;
 	if (!ffmpegCore->convertFFmpegToAudioFormat(_codecCtx->sample_fmt, &sampleFormat)) {
 		// try standard format
-		sampleFormat = asfS16;
+		sampleFormat = AUDIO_SAMPLE_FORMAT_S16;
 	}
 	if (_codecCtx->channels > 255) {
 		logger.status("Error: _codecCtx->channels > 255", "TFFmpegDecodeStream.Open");
diff --git a/mediaplugin/src/mediaplugins/ffmpeg/ffmpeg_core.cpp b/mediaplugin/src/mediaplugins/ffmpeg/ffmpeg_core.cpp
index 79f353e1..506387ba 100644
--- a/mediaplugin/src/mediaplugins/ffmpeg/ffmpeg_core.cpp
+++ b/mediaplugin/src/mediaplugins/ffmpeg/ffmpeg_core.cpp
@@ -165,20 +165,20 @@ int MediaCore_FFmpeg::findAudioStreamIndex(AVFormatContext *formatCtx) const {
 bool MediaCore_FFmpeg::convertFFmpegToAudioFormat(SampleFormat ffmpegFormat, audioSampleFormat_t *format) const {
 	switch (ffmpegFormat) {
 	case SAMPLE_FMT_U8:
-		*format = asfU8;
+		*format = AUDIO_SAMPLE_FORMAT_U8;
 		break;
 	case SAMPLE_FMT_S16:
-		*format = asfS16;
+		*format = AUDIO_SAMPLE_FORMAT_S16;
 		break;
 	case SAMPLE_FMT_S32:
-		*format = asfS32;
+		*format = AUDIO_SAMPLE_FORMAT_S32;
 		break;
 	case SAMPLE_FMT_FLT:
-		*format = asfFloat;
+		*format = AUDIO_SAMPLE_FORMAT_FLOAT;
 		break;
 #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(51,65,0)
 	case SAMPLE_FMT_DBL:
-		*format = asfDouble;
+		*format = AUDIO_SAMPLE_FORMAT_DOUBLE;
 		break;
 #endif	
 	default:
diff --git a/mediaplugin/src/mediaplugins/include/core/plugin_core.h b/mediaplugin/src/mediaplugins/include/core/plugin_core.h
index c7d078b0..69cc7f19 100644
--- a/mediaplugin/src/mediaplugins/include/core/plugin_core.h
+++ b/mediaplugin/src/mediaplugins/include/core/plugin_core.h
@@ -141,26 +141,30 @@ typedef struct pluginCore_t {
 } pluginCore_t;
 
 typedef enum audioSampleFormat_t {
-	asfUnknown,           // unknown format
-	asfU8, asfS8,         // unsigned/signed  8 bits
-	asfU16LSB, asfS16LSB, // unsigned/signed 16 bits (endianness: LSB)
-	asfU16MSB, asfS16MSB, // unsigned/signed 16 bits (endianness: MSB)
-	asfU16, asfS16,       // unsigned/signed 16 bits (endianness: System)
-	asfS32,               // signed 32 bits (endianness: System)
-	asfFloat,             // float
-	asfDouble             // double
+	AUDIO_SAMPLE_FORMAT_UNKNOWN, // unknown format
+	AUDIO_SAMPLE_FORMAT_U8,      // unsigned 8 bits
+	AUDIO_SAMPLE_FORMAT_S8,      // signed 8 bits
+	AUDIO_SAMPLE_FORMAT_U16LSB,  // unsigned 16 bits (endianness: LSB)
+	AUDIO_SAMPLE_FORMAT_S16LSB,  // signed 16 bits (endianness: LSB)
+	AUDIO_SAMPLE_FORMAT_U16MSB,  // unsigned 16 bits (endianness: MSB)
+	AUDIO_SAMPLE_FORMAT_S16MSB,  // signed 16 bits (endianness: MSB)
+	AUDIO_SAMPLE_FORMAT_U16,     // unsigned 16 bits (endianness: System)
+	AUDIO_SAMPLE_FORMAT_S16,     // signed 16 bits (endianness: System)
+	AUDIO_SAMPLE_FORMAT_S32,     // signed 32 bits (endianness: System)
+	AUDIO_SAMPLE_FORMAT_FLOAT,   // float
+	AUDIO_SAMPLE_FORMAT_DOUBLE   // double
 } audioSampleFormat_t;
 
 // Size of one sample (one channel only) in bytes
 static const int g_audioSampleSize[] = {
-	0,        // asfUnknown
-	1, 1,     // asfU8, asfS8
-	2, 2,     // asfU16LSB, asfS16LSB
-	2, 2,     // asfU16MSB, asfS16MSB
-	2, 2,     // asfU16,    asfS16
-	3,        // asfS24
-	4,        // asfS32
-	4,        // asfFloat
+	0,        // Unknown
+	1, 1,     // U8, S8
+	2, 2,     // U16LSB, S16LSB
+	2, 2,     // U16MSB, S16MSB
+	2, 2,     // U16,    S16
+	3,        // S24
+	4,        // S32
+	4,        // Float
 };
 
 typedef struct audioFormatInfo_t {
diff --git a/mediaplugin/src/mediaplugins/include/core/util.h b/mediaplugin/src/mediaplugins/include/core/util.h
index 55c5e93f..ba7ae2a4 100644
--- a/mediaplugin/src/mediaplugins/include/core/util.h
+++ b/mediaplugin/src/mediaplugins/include/core/util.h
@@ -62,7 +62,7 @@ public:
 	AudioFormatInfo() :
 		_sampleRate(0),
 		_channels(0),
-		_format(asfUnknown),
+		_format(AUDIO_SAMPLE_FORMAT_UNKNOWN),
 		_frameSize(0)
 	{}